This is a reland of 6fc6a0cbb10ee0e988b47f48935b630ba41d109d Original change's description: > Reland "Adds richer packet and ice processing to ParsedRtcEventLog." > > This is a reland of 4306a25dfcaba7defe09f5d4b669736d374fe985 > > Original change's description: > > Adds richer packet and ice processing to ParsedRtcEventLog. > > > > Bug: webrtc:10170 > > Change-Id: I0f10a8c0b5656917a806cf0f3ad88b7a6baee000 > > Reviewed-on: https://webrtc-review.googlesource.com/c/116069 > > Reviewed-by: Björn Terelius <terelius@webrtc.org> > > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#26268} > > Bug: webrtc:10170 > Change-Id: Ie523427acba02b554583223b9ef800249d8d8f2b > Reviewed-on: https://webrtc-review.googlesource.com/c/117724 > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#26350} Bug: webrtc:10170 Change-Id: I9b57ca754197822de9966ee4c93526c7f2159dfd Reviewed-on: https://webrtc-review.googlesource.com/c/118784 Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26362}
36 lines
1.4 KiB
C++
36 lines
1.4 KiB
C++
/*
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* Copyright 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "logging/rtc_event_log/logged_events.h"
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namespace webrtc {
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LoggedPacketInfo::LoggedPacketInfo(const LoggedRtpPacket& rtp,
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LoggedMediaType media_type,
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bool rtx,
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Timestamp capture_time)
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: ssrc(rtp.header.ssrc),
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stream_seq_no(rtp.header.sequenceNumber),
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size(static_cast<uint16_t>(rtp.total_length)),
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payload_type(rtp.header.payloadType),
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media_type(media_type),
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rtx(rtx),
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marker_bit(rtp.header.markerBit),
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has_transport_seq_no(rtp.header.extension.hasTransportSequenceNumber),
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transport_seq_no(static_cast<uint16_t>(
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has_transport_seq_no ? rtp.header.extension.transportSequenceNumber
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: 0)),
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capture_time(capture_time),
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log_packet_time(Timestamp::us(rtp.log_time_us())) {}
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LoggedPacketInfo::LoggedPacketInfo(const LoggedPacketInfo&) = default;
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LoggedPacketInfo::~LoggedPacketInfo() {}
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} // namespace webrtc
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